Current live broadcast for an audio/video is implemented by means of a live stream-push server. Loading of live broadcast is taken by the live stream-push server. As a result, operation loading on the server is very huge and it is not easy to process the data come from the pushed stream.
Current matured popular technology for a video communication comprises the following types.
One is WebRTC technology, which implements a web-based video meeting, and uses a standard of a WHATWG protocol. By means of providing a simple Javascript through browser, a capability for a real-time communications (RTC) is possible. The WebRTC technology provides a browser interface named “Media Stream API” to get data stream come from a camera or microphone, a browser interface named “RTC Peer Connection API” to build a stable and high efficient data flow transmission between a node to a node, as well as a browser interface named “RTC data Channel API” to build a high capacity and little delay communication channel between browsers (node to node).
A second one is popping-screen technology, wherein a popping-screen will appear in a video in real time during a period of playing the video. In such a situation, user can see popping-screens sent by himself and others when he is watching the video. During a progress of playing a video through a network, a server can get popping-screen messages from client terminals and add any popping-screen message published by respective client terminals for responding to the network-based video to the video in different timings so as to display such messages By means of such a way, client terminals are convenient to read any observation messages published by all client terminals when they are watching a video through a network.
A third one is WebSocket technology. The so-called WebSocket protocol is a new protocol used in HTML5 standard. It is possible to provide a full multiple task communication between a browser and a server, and to banefully save the resource and bandwidth used in the server so as to get a real-time communication.
A fourth one is NAT/Firewall traversal technology. The so-called Network Address Translation (“NAT” in abbreviation) is a technology that rewrites a source IP addresses and/or a destination IP addresses when IP packets are going to pass through a router or firewall. During a progress of a video conference, NAT traversal has played a vital role in a message communication between an internet and an intranet. There are two kinds of protocols for the NAT traversal. One is called STUN (Simple Traversal of UDP through NAT), UDP being a simple way to pass through NAT, the other is called TURN (Traversal Using Relay NAT), traversing NAT by means of relay. An ICE framework (Interactive Connectivity Establishment) is a comprehensive a framework for NAT traversal, and can accommodate with various NAT traversal technologies such as STUN, TURN (Traversal Using Relay NAT). ICE technology uses STUN first, trying to establish a UDP-based connection, and will use TCP if using STUN fails, by trying HTTP first and then trying HTTPS. If using TCP fails, ICE will use a relay-based TURN server. This relay-based server utilizes a Google's STUN server or a STUN server built by itself. Thus, a worst case is to use its own server for live streaming services.
For reducing the loading of the server, it is possible to utilize one of the above video communication technology to provide a browser-based audio/video live broadcast method so as to shift a portion of data processing work in the server to browser.